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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Internet Emergency Preparedness Working Group of the IETF.

	Title		: ETS Requirements for a Single Administrative Domain
	Author(s)	: K. Carlberg
	Filename	: draft-ietf-ieprep-domain-req-00.txt
	Pages		: 7
	Date		: 2004-1-12
	
This document presents a list of requirements in support of Emergency
Telecommunications Service (ETS) within a single administrative
domain. This document is an extension of the General Requirements of
[2] and focuses on a more specific set of administrative constraints
and scope.  Solutions to these requirements are not presented in this
document.

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Internet Emergency Preparedness Working Group of the IETF.

	Title		: A Framework for Supporting ETS Within a Single Administrative Domain
	Author(s)	: K. Carlberg
	Filename	: draft-ietf-ieprep-domain-frame-00.txt
	Pages		: 15
	Date		: 2004-1-12
	
This document presents a framework discussing the role of various
protocols and mechanisms that could be considered candidates for
supporting ETS within a single administrative domain.  Comments about
their potential usage as well as their current deployment are
provided to the reader.  Specific solutions are not presented.

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From exim@www1.ietf.org  Thu Jan 15 09:52:27 2004
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From: sob@harvard.edu (Scott Bradner)
Subject: [Ieprep] IEPREP in Korea
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at this point we do not plan to have an ieprep meeting in Korea 

if feel very strongly that we should please let us know

Scott & Kimberly

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 Test =)
femtsiyrotocbhc
--
Test, yep.

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muatksubuqj.exe is removed from here because it contains a virus.

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Found virus WORM_BAGLE.A in file muatksubuqj.exe
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<P>=0AHI,<BR>=0A<BR>=0AI dont know if this a relavent question in this foru=
m.<BR>=0ADoes anyone know how to implement SSL on microcontrollers?<BR>=0AO=
r <BR>=0AIs there some source code available?<BR>=0A<BR>=0ARegards,<BR>=0AA=
run<BR>=0A<BR>=0A<BR>=0A=0A</P>=0A<br><br>=0A<A target=3D"_blank" HREF=3D"h=
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HI,=0A=0AI dont know if this a relavent question in this forum.=0ADoes anyo=
ne know how to implement SSL on microcontrollers?=0AOr =0AIs there some sou=
rce code available?=0A=0ARegards,=0AArun=0A=0A=0A
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<P>=0AHi,<BR>=0A<BR>=0AI feel that there should be 2 seperate stacks(TCP/IP=
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ce. I feel the interaction should take place at a layer at TCP or above it =
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Hi,=0A=0AI feel that there should be 2 seperate stacks(TCP/IP) for split TC=
P connection.=0ABut what is the interfacing between these two stack and at =
what layer should the interfacing or interaction take place. I feel the int=
eraction should take place at a layer at TCP or above it but below the Sess=
ion layer. =0A=0AIS there any third party open source code available for TC=
P split connection.?=0A=0A=0ARegards,=0AArun=0A
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From exim@www1.ietf.org  Tue Jan 27 17:34:30 2004
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I think that, under "normal" circumstances, the QoS mechanisms for
conventional VoIP will be more than adequate to support ETS, and "special
treatment" in Call Admission Control is what will be needed for ETS.

But when  network congestion and/ or damage are severe, the "conventional"
mechanisms may no longer be adequate (especially when network failure
initiates the re-routing of Label Switched Paths, and significantly reduces
the available capacity).

As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if call
admission is working properly, traffic marking is not needed in normal
operations.  It is when there are route changes - typically due to network
failures - that marking is needed.  When there are network failures, the
capacity and route assignments on which call admission was based no longer
apply, the QoS cannot be guaranteed until the new routes are stabilized,
and the call admission process adapts to the new capacity and routes.  This
may be a very local effect, of quite short duration.  But it has a
significant impact on the call quality of the calls that are affected.
While that paper addresses an IP implementation of MLPP, many of the same
concerns apply to ETS.

Network failures, and/or  massive and sustained statistical anomalies
(large, local bursts), and the associated route changes, are the problems
we need to focus on.

It is in this context that ETS calls must be given a high probability of
meeting the QoS criteria, even when network congestion and or damage are
severe.

The requirements in draft-ietf-ieprep-domain-req-00.txt don't seem to
address this, and the framework in draft-ietf-ieprep-domain-frame-00.txt
doesn't seem to solve it.

In draft-gunn-ieprep-ip-telephony-gap-00.txt we address the  need to
support ETS SLAs to provide High Probability Toll-Quality Voice - ETS calls
must be given adequate resources through the network to ensure a high
probability that their packets suffer loss, delay, and jitter consistent
with achieving toll-quality voice even when network congestion and / or
damage are severe.

Of course, one of the conditions for such a service is that the ETS calls
are limited in terms of their relative and/or absolute use of resources.
And if the damage is so severe that there are no available paths, then
obviously they can't be given "adequate resources".  But as long as those
conditions are met,  the ETS calls should get their QoS, even when the
"normal" calls lose theirs, even if for a short time.

In addition, in order for an ETS service to be viable service (i.e., in
order for a provider to be willing to offer it at a "reasonable" cost),
there are several additional requirements.

The impact on non-ETS calls should be minimized, consistent  with providing
the ETS service.  In particular, the impact on non-ETS calls, when there
are no active ETS calls in the network, should be minimal, if any.  So any
solution that requires a lot of additional work on every packet, and/or at
every router, all the time, is not going to be viable.

There needs to be some sort of limit on the volume of ETS traffic- if ALL
the traffic using a particular resource at a particular moment in time is
ETS traffic, it becomes impossible to provide priority treatment.  Limiting
the volume of ETS traffic also reduces its attractiveness as a vehicle for
a DoS attacks (though it might make it more vulnerable to a DoS attack).

The impact on network management and administration should be minimized.
Approaches which require a lot of additional configuration and
administration, just for ETS, during normal network administration, are not
going to be viable.  Such approaches are also unlikely to be sufficiently
reliable, as there is a significant chance that undetected errors will be
made in the ETS specific configuration tasks.

Approaches which build on existing hardware, software, protocols and
procedures are far more likely to be financially viable, and far more
likely to be reliable, than approaches that rely on new or untested
protocols and techniques.

What is the appropriate way to reflect these concerns?




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Date: Tue, 27 Jan 2004 16:35:49 -1000
To: Janet P Gunn <jgunn6@csc.com>
From: Fred Baker <fred@cisco.com>
Subject: Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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At 12:32 PM 1/27/2004, Janet P Gunn wrote:
>As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if call 
>admission is working properly, traffic marking is not needed in normal 
>operations.

lest I be misunderstood, let me restate that. What I wrote was:

    One will ask the value of the multiple DSCPs. They are, in fact, of
    limited to no value in normal operation, as all vice and video
    traffic should have been admitted, and therefore capacity will have
    been assigned to them. The behavior of this service will be
    indistinguishable from the EF PHB regardless of traffic marking.

In retrospect, I wonder what the vice traffic would be :^)

This is not to be understood as saying "no traffic marking is required"; 
one still needs RFC 3246. But if bandwidth admission is operating 
correctly, having N different marks will be operationally indistinguishable 
from having a single EF mark, as the total number of bytes on the wire will 
be the same.

>It is in this context that ETS calls must be given a high probability of 
>meeting the QoS criteria, even when network congestion and or damage are 
>severe.

I wonder if we are communicating here. RFC 3246 is designed to operate in a 
congested network, and to operate in a network in which routing may change. 
The comment in the mlpp-that-works draft - which if it is giving cause for 
confusion I will happily remove - is basically me bending over backwards to 
be intellectually honest and see if I can find a case where CAC may be 
insufficient.

Think about this for a moment: the argument being placed for MLEF is that 
even in the presence of some loss, voice is intelligible. If that is true, 
then just after a route change, during the brief interval between the data 
flow being switched to a new path and the CAC following that and either 
OKing the flow or deciding to shut down the call, when there may be some 
loss, voice should be equally intelligible, no?

Or does the argument that voice is able to work around a modest level of 
loss only work when it comes from the proponents of multiple code points? 
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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: "'Janet P Gunn'" <jgunn6@csc.com>
Cc: <ieprep@ietf.org>
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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Janet,

The following response may be a bit picky on some issues/comments, but I
feel it will be helpful to clarify things a bit.
 
> I think that, under "normal" circumstances, the QoS mechanisms for
> conventional VoIP will be more than adequate to support ETS, and
"special
> treatment" in Call Admission Control is what will be needed for ETS.

Within the context of what we have written in the past in the IETF, ETS
is an umbrella term that is meant to cover a wide variety of
applications.  Some of these applications (eg, I-Am-Alive) do not
support the concept of CAC. So to say that "special treatment" (whatever
that is) of CAC is what will be needed for ETS is rather too specific a
statement to make.  

I think we can agree that in those applications used to support ETS and
dependent on CAC, an ability to distinguish and perform CAC is of high
interest and needs to be supported.
 
> As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if
call
> admission is working properly, traffic marking is not needed in normal
> operations.  It is when there are route changes - typically due to
network
> failures - that marking is needed.  

Keep in mind that Fred's draft is in respect to MLPP.  Arguments from
the draft may be applied to other circumstances, but I would refrain
from treating it as a blanket statement for all applications and
situations.  I've worked on systems in the past where "marking" was used
to separate traffic into either different physical subnets or different
virtuals paths -- neither case where the route changes existed because
of network failures.

> When there are network failures, the
> capacity and route assignments on which call admission was based no
longer
> apply, the QoS cannot be guaranteed until the new routes are
stabilized,
> and the call admission process adapts to the new capacity and routes.
> This may be a very local effect, of quite short duration.  But it has
a
> significant impact on the call quality of the calls that are affected.
> While that paper addresses an IP implementation of MLPP, many of the
same
> concerns apply to ETS.

Ok

> Network failures, and/or  massive and sustained statistical anomalies
> (large, local bursts), and the associated route changes, are the
problems
> we need to focus on.
> 
> It is in this context that ETS calls must be given a high probability
of
> meeting the QoS criteria, even when network congestion and or damage
are
> severe.

Ok.

> The requirements in draft-ietf-ieprep-domain-req-00.txt don't seem to
> address this, and the framework in
draft-ietf-ieprep-domain-frame-00.txt
> doesn't seem to solve it.

Actually, that's fine.  The former draft is not aimed at a specific
application or system (this is stated in the draft).  It is admittedly
broad and acts as a foundation for the context of a single domain.  The
latter draft is not meant to offer solutions, but rather to discuss
applicable and related protocols that one *could* encounter within the
specific scope of a single domain.  I will expand this below where you
have a specific question. 

> In draft-gunn-ieprep-ip-telephony-gap-00.txt we address the  need to
> support ETS SLAs to provide High Probability Toll-Quality Voice - ETS
> calls
> must be given adequate resources through the network to ensure a high
> probability that their packets suffer loss, delay, and jitter
consistent
> with achieving toll-quality voice even when network congestion and /
or
> damage are severe.

<snip>
 
> What is the appropriate way to reflect these concerns?

Through a draft (and I state it with no intention of being obnoxiously
obvious). If your question is, how does one deal with the three
seemingly related drafts?
   1. draft-ietf-ieprep-domain-req-00.txt, 
   2. draft-ietf-ieprep-domain-frame-00.txt, and 
   3. draft-gunn-ieprep-ip-telephony-gap-00.txt, 

then I would follow the position I stated above and have (3) simply be a
more specific continuation of what has been stated in (1) and (2) above.
In my view it may be more constructive to be very specific and identify
specific systems/efforts like GETS or NS/EP or WPS.  If the chairs or AD
feel differently, I'm sure they'll chime in.

Cheers,

-ken


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If the offered load exceeds the network capacity, some packets will be
dropped.  If a line is at
95% of capacity, a 10% surge would cause 5% of the packets to be
dropped, degrading most of the conversations.  ( I am simplifying here
but I think this is clear.)

The benefit of multiple DSCPs is that the sessions designated most
critical are unaffected. Actual experiment has shown that this does
work.

In draft-baker-tsvwg-mlpp-that-works.00 it is assumed that the CAC
will ensure that voice is not overloaded.
Two problems with this are:

	1)  If silence suppression is used and the bandwidth "oversubscribed"
(standard practice) then sometimes, more people on one end will talk
and cause congestion.

	2)  If the area controlled by the CAC is large, propagation delay
will limit the reaction time of the system and the computation to
oversee the entire theatre may also be a limiting factor.  This may
cause short congestion incidents (several seconds long) which could be
mitigated by MLEF.  I'm not saying you don't also need the CAC.  I'm
saying
you need something local and fast reacting (at the packet level rather
than the call level) to ensure that the high priority packets get
thru.

Steve Silverman



> -----Original Message-----
> From: ieprep-admin@ietf.org
> [mailto:ieprep-admin@ietf.org]On Behalf Of
> Fred Baker
> Sent: Tuesday, January 27, 2004 9:36 PM
> To: Janet P Gunn
> Cc: ieprep@ietf.org
> Subject: Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
>
>
> At 12:32 PM 1/27/2004, Janet P Gunn wrote:
> >As Fred Baker says in
> draft-baker-tsvwg-mlpp-that-works.00.txt, if call
> >admission is working properly, traffic marking is not
> needed in normal
> >operations.
>
> lest I be misunderstood, let me restate that. What I wrote was:
>
>     One will ask the value of the multiple DSCPs. They are,
> in fact, of
>     limited to no value in normal operation, as all vice and video
>     traffic should have been admitted, and therefore
> capacity will have
>     been assigned to them. The behavior of this service will be
>     indistinguishable from the EF PHB regardless of traffic marking.
>
> In retrospect, I wonder what the vice traffic would be :^)
>
> This is not to be understood as saying "no traffic marking
> is required";
> one still needs RFC 3246. But if bandwidth admission is operating
> correctly, having N different marks will be operationally
> indistinguishable
> from having a single EF mark, as the total number of bytes
> on the wire will
> be the same.
>
> >It is in this context that ETS calls must be given a high
> probability of
> >meeting the QoS criteria, even when network congestion and
> or damage are
> >severe.
>
> I wonder if we are communicating here. RFC 3246 is designed
> to operate in a
> congested network, and to operate in a network in which
> routing may change.
> The comment in the mlpp-that-works draft - which if it is
> giving cause for
> confusion I will happily remove - is basically me bending
> over backwards to
> be intellectually honest and see if I can find a case where
> CAC may be
> insufficient.
>
> Think about this for a moment: the argument being placed
> for MLEF is that
> even in the presence of some loss, voice is intelligible.
> If that is true,
> then just after a route change, during the brief interval
> between the data
> flow being switched to a new path and the CAC following
> that and either
> OKing the flow or deciding to shut down the call, when
> there may be some
> loss, voice should be equally intelligible, no?
>
> Or does the argument that voice is able to work around a
> modest level of
> loss only work when it comes from the proponents of
> multiple code points?



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        "Janet P Gunn" <jgunn6@csc.com>, <ieprep@ietf.org>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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comments below

At 11:33 AM 1/29/2004 -0500, Steve Silverman wrote:
>If the offered load exceeds the network capacity, some packets will be
>dropped.  If a line is at
>95% of capacity, a 10% surge would cause 5% of the packets to be
>dropped, degrading most of the conversations.  ( I am simplifying here
>but I think this is clear.)
>
>The benefit of multiple DSCPs is that the sessions designated most
>critical are unaffected.

without active feedback to those lower priority sessions....

>Actual experiment has shown that this does
>work.

yes it does - but without regard to any call quality.

If the surge is 25 or 50 or 75% above maximum load possible, then what 
happens to the lower priority calls? There isn't any feedback to the users 
(other than pure silence). Further, there isn't any feedback (signaling) to 
their devices what's happening or to hang up the call. There's nothing. How 
long is this expected to last for? No one knows.

Further, no one can test the system (because of no feedback).

No one can point to where there might be potential problems with this 
architecture either (is the system behaving correctly?). There is no 
feedback to discover or monitor its performance. Where is it supposed to be 
discovered that a link between any two routers isn't fast enough (because 
there are perhaps frequent silence periods)?


>In draft-baker-tsvwg-mlpp-that-works.00 it is assumed that the CAC
>will ensure that voice is not overloaded.
>Two problems with this are:
>
>         1)  If silence suppression is used and the bandwidth "oversubscribed"
>(standard practice) then sometimes, more people on one end will talk
>and cause congestion.
>
>         2)  If the area controlled by the CAC is large, propagation delay
>will limit the reaction time of the system

where's the propagation delay? the surge won't happen without the signaling 
having already occurred through the congestion point; meaning the system 
(read: routers) already know about all the call (flow) attempts - even when 
there is too much offered load for an interface. The router can either say 
"sorry, not enough resources available to complete your call attempt" or 
"sorry, but there was another call with higher priority, and I have to shut 
your call down now". BTW - these are device signals, this isn't a .wav file 
playing to the user or anything.

>and the computation to
>oversee the entire theatre may also be a limiting factor.

see above, computation occurs at the router that is experiencing this 
surge. It is in charge of the packet forwarding. It is aware of all the 
existing flows it has accepted. It is in charge of accepting new flows and 
preempting lesser priority ones if multilevel offered load exceeds the 
maximum load possible.

>This may
>cause short congestion incidents (several seconds long) which could be
>mitigated by MLEF.  I'm not saying you don't also need the CAC.  I'm
>saying
>you need something local and fast reacting

In the router itself isn't fast enough...?

>(at the packet level rather
>than the call level)

packet level = Layer 3

Call level = what? Layer 4, 5 or 7?

>to ensure that the high priority packets get
>thru.

I am a believer in high priority calls getting through, and for all calls 
that remain up - having all their packets arrive safely too.


>Steve Silverman


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: "'James M. Polk'" <jmpolk@cisco.com>
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James,

> If the surge is 25 or 50 or 75% above maximum load possible, then what
> happens to the lower priority calls? There isn't any feedback to the
users
> (other than pure silence). Further, there isn't any feedback
(signaling)
> to
> their devices what's happening or to hang up the call. There's
nothing.
> How long is this expected to last for? No one knows.
> 
> Further, no one can test the system (because of no feedback).

I'm sorry if I missed an argument that may be in one of the related
tsvwg drafts, but is there a specific form of the feedback you are
looking for?  We have RTCP reports that can provide a measure of
feedback on an end-to-end basis.  Admittedly, these are generally not
provided to the user and they are not targeted towards a specific
application or underlying network service, but the hooks are there to at
least inform the users of loss versus silence.

-ken


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Comments below

At 04:26 PM 1/29/2004 -0500, Ken Carlberg wrote:
>James,
>
> > If the surge is 25 or 50 or 75% above maximum load possible, then what
> > happens to the lower priority calls? There isn't any feedback to the
>users
> > (other than pure silence). Further, there isn't any feedback
>(signaling)
> > to
> > their devices what's happening or to hang up the call. There's
>nothing.
> > How long is this expected to last for? No one knows.
> >
> > Further, no one can test the system (because of no feedback).
>
>I'm sorry if I missed an argument that may be in one of the related
>tsvwg drafts, but is there a specific form of the feedback you are
>looking for?

This thread is based on:

http://www.ietf.org/internet-drafts/draft-silverman-diffserv-mlefphb-02

and the response in both

http://www.ietf.org/internet-drafts/draft-baker-mlef-concerns-00.txt, and
http://www.ietf.org/internet-drafts/draft-baker-mlpp-that-works-00.txt

This isn't a direct ETS topic - but it is near the topic, and because of 
recent conversations on the list and elsewhere that infer MLEF is a 
replacement for CAC.

Both the bottom two contend that in an MLPP service, there must be a 
feedback mechanism informing the endpoints if there has been a preemption 
of resources. The MLEF ID contends that lower priority packets should be 
vulnerable to packet loss in times of contention without regard to the 
impact on voice quality for an entire precedence level of calls through 
that congested interface.

We have demonstrated in lab tests to DISA and various other organizations 
that, a 1%, 5% and higher packet loss has significant impact on Voice 
Quality (VQ) of all codecs, while at the same time - having some 
organizations maintain the requirement that VQ cannot drop below a 4.0 MOS.


>We have RTCP reports that can provide a measure of
>feedback on an end-to-end basis.

Aren't these for after a call, and propagated up to the requesting NM station?

>Admittedly, these are generally not
>provided to the user

One of the primary MLPP Service requirements is to have a tone or 
announcement to all affected users if there is preemption at the endpoint 
or in the network.

>and they are not targeted towards a specific
>application or underlying network service, but the hooks are there to at
>least inform the users of loss versus silence.

It is Fred Baker's and my opinion that an In_band soft_state control plane 
would be the very best of all worlds to provide all manners of offered load 
to capacity analysis at the congestion point - and not continue to rely on 
some server architecture to maintain the congestive state of all interfaces 
for 1000s of router interfaces (in rear-time).


>-ken
>
>
>_______________________________________________
>Ieprep mailing list
>Ieprep@ietf.org
>https://www1.ietf.org/mailman/listinfo/ieprep


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: "'James M. Polk'" <jmpolk@cisco.com>
Cc: <ieprep@ietf.org>
Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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Thanks for the recap.

> >We have RTCP reports that can provide a measure of
> >feedback on an end-to-end basis.
> 
> Aren't these for after a call, and propagated up to the requesting NM
> station?

The reports are sent periodically throughout the session between end
points.  Implementors make their own choices if and when to display the
stats.  At UCL, the RAT (VoIP) tool allows the user to see the stats
updated in real time.  For other implementations, your mileage may very.
 
-ken



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>
> This thread is based on:
>
> http://www.ietf.org/internet-drafts/draft-silverman-diffserv
> -mlefphb-02
>
> and the response in both
>
> http://www.ietf.org/internet-drafts/draft-baker-mlef-concern
> s-00.txt, and
> http://www.ietf.org/internet-drafts/draft-baker-mlpp-that-wo
> rks-00.txt
>
> This isn't a direct ETS topic - but it is near the topic,
> and because of
> recent conversations on the list and elsewhere that infer MLEF is a
> replacement for CAC.

We are not saying MLEF is a replacement for CAC, but a compliment.  I
don't think that a CAC alone can meet the DOD requirements but I think
that
a CAC and MLEF can meet those requirements (or most of them).

>
> Both the bottom two contend that in an MLPP service, there
> must be a
> feedback mechanism informing the endpoints if there has
> been a preemption
> of resources.

I agree with this but doubt that the CAC can do this job in real time.

The MLEF ID contends that lower priority
> packets should be
> vulnerable to packet loss in times of contention without
> regard to the
> impact on voice quality for an entire precedence level of
> calls through
> that congested interface.

I believe that in the event of an overload, this is unavoidable even
if only for a short time.


...

>
> It is Fred Baker's and my opinion that an In_band
> soft_state control plane
> would be the very best of all worlds to provide all manners
> of offered load
> to capacity analysis at the congestion point - and not
> continue to rely on
> some server architecture to maintain the congestive state
> of all interfaces
> for 1000s of router interfaces (in real-time).

James:  Have you published anything on this idea?


Steve



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Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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> We have demonstrated in lab tests to DISA and various other 
> organizations 
> that, a 1%, 5% and higher packet loss has significant impact on Voice 
> Quality (VQ) of all codecs, while at the same time - having some 
> organizations maintain the requirement that VQ cannot drop 
> below a 4.0 MOS.

I don't think the 4.0 MOS requirement is applicable to all (perhaps
most) voip carrying networks that could benefit from MLEF or another
packet level marking scheme.  This is cited as a requirement for some
DoD switched networks and does not necessarily extend to ALL voip
carrying networks, or all DoD voip carrying networks, or even all DoD
switched voice networks.  Actually, providing a 3.0 MOS would be a
significant improvement to some current voice networks - especially
wireless, mobile, low bandwidth ones.  Given this, 30% packet loss is
sustainable and acceptable in some situations.  Furthermore, MOS is not
necessarily the best measure of voice quality for all voice networks.
For example, in tactical networks, intelligibility is important while
fidelity is not as important.

Regardless, I realize if someone said that some arbitrary X quality was
required, you could come with the same argument with different packet
loss numbers.  The point is, each network and even each link may have
different requirements.  Providing multiple tools (e.g. RSVP based CAC +
MLEF) to engineer each network, or in some cases each link, for the
specific requirements is needed.

Chris

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